US8355394B2 - System and method for dynamic telephony resource allocation between premise and hosted facilities - Google Patents
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/50—Centralised arrangements for answering calls; Centralised arrangements for recording messages for absent or busy subscribers ; Centralised arrangements for recording messages
- H04M3/51—Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing
- H04M3/523—Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing with call distribution or queueing
- H04M3/5237—Interconnection arrangements between ACD systems
Definitions
- the present invention relates to telecommunication and a networked computer telephony system including the Internet and the Public Switched Telephone System, and more particularly to user-configurable allocation of telephony resources between a hosted facility and a subscriber's premise.
- the first is a network of telephone systems in the form of the Public Switched Telephone System (PSTN). This network was initially designed to carry voice communication, but later also adapted to transport data.
- PSTN Public Switched Telephone System
- the second is a network of computer systems in the form of the Internet.
- the Internet has been designed to carry data but also increasingly being used to transport voice and multimedia information.
- Computers implementing telephony applications have been integrated into both of these telecommunication networks to provide enhanced communication services. For example on the PSTN, computer telephony integration has provided more functions and control to the POTS (Plain Old Telephone Services).
- POTS Personal Telephone Services
- FIG. 1A illustrates a typical configuration of a conventional computer telephony server operating with a Public Switched Telephone Network (PSTN) and/or the Internet.
- PSTN Public Switched Telephone Network
- the PSTN 10 includes a network of interconnected local exchanges or switches 12 . Around each exchange is provisioned a cluster of telephone lines to which telephones, modems, facsimile machines may be attached. Other private exchanges such as Private Brach Exchange PBX 20 may also be connected to the PSTN to form a public/private telephone network. Voice or data is carried from a source node to a destination node on the network by establishing a circuit path along the PSTN effected by appropriately switching the interconnecting exchanges.
- PBX 20 Private Brach Exchange
- the point-to-point transmission is therefore circuit-switched, synchronous and using a dedicated channel of fixed bandwidth (64 kbs).
- the exchanges have mostly been upgraded to handle digital, time-division multiplexed trunk traffic between the exchanges.
- External digital communication systems typically communicate with the PSTN by interfacing with an exchange such as 12 .
- a common digital interface at the exchange is PRI (Primary Rate Interface) which is part of an ISDN (Integrated Services Digital Network) and is usually provided by a T 1 or E 1 trunk line.
- PRI Primary Rate Interface
- ISDN Integrated Services Digital Network
- the Internet 30 is a worldwide interconnection of IP (Internet Protocol) networks, with interconnecting computers communicating with each other using TCP/IP (Transmission Control Protocol/Internet Protocol). Some of the computers may also be interconnected by a private segment of the IP network with restricted access.
- IP Internet Protocol
- TCP/IP Transmission Control Protocol/Internet Protocol
- Some of the computers may also be interconnected by a private segment of the IP network with restricted access.
- IP network data from a source node is cast into a number of packets that may individually be transported via multiple paths on the network to be reassembled at a destination node.
- the transmission on the IP network is packet-switched and asynchronous.
- VoIP Internet Protocol
- the VoIP protocol includes a number of standards.
- one such standard is the H.323 standard promulgated by the ITU (International Telecommunication Union) aims to ensure VoIP interoperability. It provides a specification for communication of multimedia such as voice, data and video between terminal equipment over IP networks.
- the terminal equipment communicating on the Internet includes personal computers with telephony capabilities 40 , VoIP phones 42 that can connect to the Internet directly, and other networked telephony appliances.
- Web applications 44 in general and web pages in particular are written in HTML (HyperText Markup Language) and are hosted by web servers 46 on the Internet. Each web page can be called up by its URL (Uniform Resource Locator), which can translate to an IP address on the Internet.
- URL Uniform Resource Locator
- These web pages may be requested and processed by a web browser running on a computer connected to the Internet. The web browser retrieves the web page under HTTP (HyperText Transfer Protocol) and parses the HTML codes on the web page to execute it. Typically, the execution of HTML codes on a web page results in rendering it into a display page on the browser or client computer.
- HTTP HyperText Transfer Protocol
- HTML HyperText Markup Language
- CTI Computer Telephony Integration
- a computer When applied to a phone or a terminal equipment, CTI provides added features to an end user's phone.
- CTI is usually implemented with a CT (Computer Telephony) server, such as CT server 50 .
- CT server 50 executes telephony applications that can provide custom services such as interactive voice response, customer service or help desk for an organization.
- the CT server 50 can be configured to interface via a PSTN interface 52 with an exchange 12 to receive and process calls pertaining to a predefined set of telephone numbers on the PSTN.
- the CT server 50 can also be configured to interface via an IP network interface 54 with the Internet to receive and process calls pertaining to a predefined set of telephone numbers or IP addresses.
- the CT server 50 is usually a computer operating under UNIX or Microsoft Windows NT and is running installed customized application software 56 for the various voice applications.
- the CT server provides a set of API 58 (Application Program Interface) which is a set of procedures, protocols and tools for building software applications. These APIs are generally proprietary and specific to the individual hardware manufacturers. Developing an application on an existing CT server would involve a highly specialized application developer undertaking a fairly complex task of coding the application in C++ or JAVA, programming language and employing and invoking the APIs specific to the hardware.
- U.S. Pat. No. 6,011,844 discloses a distributed call center system in which a business call center running a custom interactive voice response application is essentially replicated in a number of local points of presence to reduce communication cost when connecting a local customer.
- FIG. 1B illustrates a Point-Of-Presence call center management system disclosed in U.S. Pat. No. 6,011,844.
- the system is designed to minimize long distance toll call when a customer 70 is calling a business call center 60 .
- the business call center typically runs a customized interactive voice response application 66 that implements a complete business solution to answer, service, queue and route inbound customer calls.
- the customer 70 at a local exchange 72 will in general be calling long distance to the business call center 60 that is local to a remote exchange 74 .
- the customer requests to speak to a live agent 68 at the business call center his or her call is queued until an agent is available.
- the POP call center management system deploys a number of POP call centers 80 across the Public Switched Telephone Network (PSTN) 10 so that a customer's call at a local exchange 72 is intercepted at a local POP call center 80 .
- PSTN Public Switched Telephone Network
- Each POP call center essentially serves as a local-presence business call center except without the live agent. This is accomplished by having each POP call center executing the application such as 66 ′, 66 ′′ locally.
- the local applications 66 ′, 66 ′′ can be a full replica of the application 66 residing at the business call center or they can be partial ones with some of the resources such as voice prompts, menus, etc., being accessed dynamically from the application 66 as needed.
- the application 66 that resides at the business call center is accessible by the POP call centers via an interconnecting virtual private network 90 .
- HTML or XML may be used when the POP call center access conveniently packaged units of information or application from the business call center across the call center virtual private network 90 .
- the customer's call is basically handled at a POP call center local to the customer.
- a queue is set up at the business call center until an agent becomes available. Only then will the POP call center convert the customer's local call to a long distance call to the business call center.
- the call traffic for the interactive voice response portion is carried between the local exchange 72 and a POP call center 80 .
- the call traffic between the customer 70 and a live agent 68 is carried via a long distance portion 76 of PSTN, or in other disclosed embodiments, over the call center virtual private network 90 or the Internet 30 .
- Prior computer telephony systems have infrastructures that do no allow easy development and deployment of telephony applications.
- the system illustrated in FIG. 1A requires the telephony application to be hosted in a call center type of telephony server and requires specialized knowledge of the telephony hardware to develop telephony applications.
- the same is true for the system illustrated in FIG. 1B with the variation that the call center is effectively replicated at various local points of presence on the global telephone network.
- a network system and method allows sharing of resources between voice application gateway centers (“voice centers”) operated by a hosted facility and by a subscriber or customer on premise.
- voice centers voice application gateway centers
- a customer is able to configure the network system to have some calls processed on premise and some calls processed by the host facility.
- a hosted facility is able to configure the network system to have some calls originally to be processed by the hosted facility to be processed by an available premise voice center.
- the invention is accomplished by a semi-real-time manipulation of a voice-center directory by which a destination voice center is selected to process a call.
- At least a provisioning management server with a user interface for a user to easily update lookup information in the voice-center directory.
- the lookup information enables a voice center to be selected as a function of the dialed number as well as routing rules.
- the user either an operator from the hosted facility or a customer with premise equipment, is able to update the routing rules in the voice-center directory.
- the voice-center selection function of the voice-center directory also depends on the degree of availability or current capacity of individual voice centers. This is accomplished by having the individual voice centers updating the voice-center directory regarding their current capacity at predetermined time intervals.
- the network of telephony system includes a voice center receiving a routed call and retrieving a voice application associated with the dialed number and appropriate for that voice center.
- the call can potentially be routed to any one of a number of voice centers, preferably there is a version of the voice application appropriate for each of the number of voice centers.
- a local network address of the appropriate version of the voice application is cached in each voice center so that the voice center can use it to retrieve the voice application without having to perform a look up in an external directory.
- premise customers can commit to install on premise a predefined amount of resources without over budgeting and yet be able to overflow and failover to hosted resource for unplanned call volume growth, peak call seasons, and disaster recovery plans.
- the hybrid resource allocation system described also offers the following business advantages. It allows a service provider to sell premise-based telephony software and hosted telephony services as a combined product/service, such that the customer's premise telephony software ports integrate seamlessly with the service provider's hosted telephony software ports.
- the customer will be allowed to use an online management console to configure and manage his premise-based software, making it possible for him to set parameters for how many calls he wants the premise system (as opposed to the hosted system) to take, under what circumstances to route calls through the hosted or premise system.
- FIG. 1A illustrates a typical configuration of a conventional computer telephony server operating with a Public Switched Telephone Network (PSTN) and/or the Internet.
- PSTN Public Switched Telephone Network
- FIG. 1B illustrates a Point-Of-Presence call center management system disclosed in U.S. Pat. No. 6,011,844.
- FIG. 2 illustrates an Application Gateway Center (vAGC) for processing telephony applications on the Internet and the PSTN, according to a general scheme of the present invention.
- vAGC Application Gateway Center
- FIG. 3 is a flow diagram illustrating the setup for provisioning and processing voice applications according to a general embodiment of the present invention.
- FIG. 4A illustrates a preferred configuration of the inventive system with respect to the Internet and the PSTN.
- FIG. 4B is a flow diagram illustrating an exemplary call routing and processing in the preferred configuration shown in FIG. 4A .
- FIG. 5 illustrates an alternative preferred configuration of the inventive system with respect to the Internet and the PSTN.
- FIG. 6 is a block diagram illustrating the components of the Application Gateway Center.
- FIG. 7 is a block diagram illustrating schematically the components of the media conversion proxy server.
- FIG. 8 is a detailed block diagram of the Application Gateway Server, which is the main component of the Application Gateway Center.
- FIG. 9 is a system block diagram of a network traffic monitoring system operating in cooperation with the Distributed Application Telephony Network System of the present invention.
- FIG. 10 illustrates a preferred network configuration including the PSTN and the Internet for practicing the invention.
- FIG. 11 is a block diagram of the provisioning management server.
- FIG. 12 illustrates the information contained in an entry in the voice-center directory.
- FIG. 13 illustrates another example of a voice center updating its current capacity or availability information.
- FIG. 14 illustrates an example of user configurable call routing between voice centers located on different sites.
- FIG. 15 illustrates another example of user configurable call routing using the vAGC routing proxy instead of the access server.
- FIG. 16 illustrates maintaining different versions of voice applications for a given dialed number.
- FIG. 17 is a flow diagram illustration the scheme for allocation of voice service and resource in a hybrid environment of premise and hosted sites.
- FIGS. 18A-18J illustrate embodiments of the routing rules of STEP 740 in FIG. 17 .
- the Internet is a worldwide network of IP networks communicating under TCP/IP.
- voice and other multimedia information are transported on the Internet under the VoIP (Voice-over-IP) protocol, and under the H.323 standard that has been put forward for interoperability.
- VoIP Voice-over-IP
- H.323 H.323 standard that has been put forward for interoperability.
- SIP Session Initiation Protocol
- FIG. 2 shows a typical environment including the Internet and the PSTN in which the present invention is practiced.
- the Internet 30 acts as a VoIP network for communication between terminal equipments, such as personal computers (PC) 40 with telephony capabilities and/or dedicated VoIP phones 42 connectable directly to the Internet.
- terminal equipments such as personal computers (PC) 40 with telephony capabilities and/or dedicated VoIP phones 42 connectable directly to the Internet.
- PC personal computers
- Each terminal equipment on the Internet has an IP address that may also be associated with a predefined call number so that one terminal equipment may call another one by its IP address or equivalently by its call number.
- HTML applications such as an application 44 hosted on a web server 46 that may also interact with other clients and servers 48 on the Internet.
- the PSTN 10 is a network of exchanges. Each exchange is provisioned with a plurality of telephone lines or nodes having designated call numbers. Two PSTN nodes are connectable by switching the intervening exchanges to form a circuit.
- the PSTN and the Internet are interconnected by means of access servers such as an access server 14 .
- This enables communication between a PSTN node and an Internet node.
- a telephonic call transported between two network nodes comprises a signaling portion for setting up and tearing down the call and a media portion for carrying the voice or multimedia data.
- the access server 14 essentially converts both of these portions to an appropriate format across the interface between the two types of networks.
- On the PSTN side the digital interface is PRI and on the Internet side the interface is VoIP.
- a wireless or mobile telephone network may similarly be considered as an extension of the PSTN. It is typically connected to the PSTN via a suitable interface implemented by a gateway.
- FIG. 2 illustrates an Application Gateway Center (vAGC) for processing telephony applications on the Internet and the PSTN, according to a general scheme of the present invention.
- the Application Gateway Center (vAGC) 100 is a voice-processing center on the Internet 30 for intercepting and processing calls to anyone of a set of designated telephone call numbers. The calls may originate or terminate on any number of interconnected telecommunication networks including the Internet 30 , the PSTN 10 , and others (not shown) such as wireless networks.
- the vAGC 100 processes each call according to the telephony application (vAPP) associated with the called number.
- vAPP telephony application
- a plurality of these associated telephony applications, vAPPs such as 110 , . . .
- XML applications denoted more specifically as (vXML) applications
- vXML applications are coded in XML scripts that also contain custom telephony XML tags.
- the vXML scripts allow complete telephony applications to be coded.
- the set of designated call numbers handled by the vAGC 100 are registered in a directory, such as DIR 0 .
- a directory such as DIR 0 .
- the plurality of telephony applications vAPP 110 , . . . , 112 each associated with at least one designated call number, is accessible by the vAGC from the Internet.
- Each application is coded in vXML and is being hosted as a webpage on a web server on the Internet.
- a directory DIR 1 provides the network address of the various applications.
- the vAGC 100 uses the call number (or dialed number DN) to look up DIR 1 for the location/address (whether a URL or an IP address or some other location method) of the vAPP associated with the DN.
- the vAGC 100 retrieves the vXML webpage and executes the call according to the vXML scripts.
- FIG. 3 is a flow diagram illustrating the setup for provisioning and processing voice applications according to a general embodiment of the present invention. Provisioning of a designated call number with its associated vAPP is described in steps 130 , 132 and 134 .
- Step 130 For a given call number DN, create an associated telephony application, vAPP in vXML, and deploy it on the Internet with a specific IP address or URL.
- Step 132 Provide any media, files and web applications that are requested or act on by vAPP.
- Step 134 Update the directory DIR 1 so that the address of vAPP can be obtained by querying with its associated call number DN.
- Step 140 vAGC receives a call with DN routed thereto.
- Step 142 vAGC uses DN to look up DIR 1 for the address of the webpage for vAPP.
- Step 144 vAGC requests and retrieves the webpage containing vXML scripts for vAPP.
- Step 146 vAGC processes the call according to the retrieved vXML scripts for vAPP.
- FIG. 4A illustrates a preferred configuration of the inventive system with respect to the Internet and the PSTN.
- the configuration is similar to that shown in FIG. 2 except there is a plurality of Application Gateway Centers (vAGCs) 100 , 100 ′, . . . , 100 ′′ deployed on the Internet 30 .
- vAGCs Application Gateway Centers
- This will provide redundancy, capacity and load-balancing for executing the plurality of telephony applications vAPP 110 , . . . , 110 ′′ being hosted by web servers 112 , . . . , 112 ′ on the Internet.
- LECs Local Exchange Carriers
- AS access server
- Each access server communicates on the one hand with an exchange of the LEC via the PRI interface and on the other hand with the Internet via the VoIP protocol such as H.323 or SIP.
- the VoIP protocol such as H.323 or SIP.
- FIG. 4B is a flow diagram illustrating an exemplary call routing and processing in the preferred configuration shown in FIG. 4A .
- the numeral in parenthesis denotes the route taken.
- a new call originates from a telephone line 11 on a local exchange. Since the call is made to a dialed number (DN) registered as one of the numbers handled by the vAGC, it is routed to a vAGC such as vAGC 100 after a lookup from DIR 0 .
- the vAGC 100 initiates a new session for the call and looks up DIR 1 for the net address of the telephony application vAPP 110 associated with the DN.
- the vAGC 100 retrieves vAPP 110 and proceeds to process the vXML scripts of vAPP 110 .
- the vXML scripts dictate that the new call is to be effectively routed back to the PSTN to a telephone 13 on another local exchange. In another example, the vXML scripts dictate that the call is to be effectively routed to a VoIP phone 15 on the Internet.
- the vAGC creates separate sessions for the two nodes and then bridges or conferences them together. This general scheme allows conferencing between multiple parties.
- the vXML scripts allows the call to interact with other HTML applications or other backend databases to perform on-line transactions.
- the present system allows very power yet simple telephony applications to be built and deployed on the Internet.
- the following are some examples of the vAPP telephony applications contemplated.
- a “Follow me, find me” application sequentially calls a series of telephone numbers as specified by a user until one of the numbers answers and then connects the call. Otherwise, it does something else such as takes a message or sends e-mail or sends the call to a voice center, etc.
- a Telephonic Polling application looks up from a database the telephone numbers of a population to be polled.
- a Help Desk application plays a series of interactive voice prompts/messages in response to the called party's responses and possibly connects the call to a live agent as one option, etc.
- a Stock or Bank Transactions application plays a series of interactive voice prompts/messages in response to the called party's responses and conducts appropriate transactions with a backend database or web application, etc.
- FIG. 5 illustrates an alternative preferred configuration of the inventive system with respect to the Internet 30 and the PSTN 10 .
- the arrangement is similar to that of FIG. 4A except at individual LECs, the Application Gateway Centers vAGC 100 , 100 ′, . . . , 100 ′′ are respectively co-located with the local access servers AS 14 ′′, 14 ′, . . . , 14 .
- This configuration provides higher quality-of-service (QoS) at the expense of repeating the vAGC at every LEC.
- QoS quality-of-service
- FIG. 6 is a block diagram illustrating the components of the Application Gateway Center.
- the Application Gateway Center vAGC 100 may be considered to be a facility hosting a cluster of servers for the purpose of receiving calls and running the associated telephony applications, vAPPs, reliably and efficiently.
- the vAGC 100 comprises two IP network segments.
- An Internet network segment 130 connects the vAGC 100 to the Internet.
- a local IP network segment 140 allows direct communication between an application gateway server 200 , a cache server 310 and a media conversion proxy server 320 .
- the cache server 310 and the media conversion proxy server 320 are also connected directly to the Internet via the Internet network segment 130 .
- multiple servers of each type are installed in the vAGC 100 .
- the application gateway server 200 exchanges data with the Internet indirectly through the cache server 310 and possibly the media conversion proxy server 320 .
- the AGS 200 retrieves the associated vAPP from a website and proceeds to execute the vXML scripts of the vAPP.
- associated media and/or files may also be retrieved from various sites as part of the vAPP suite.
- the vXML scripts, media and files that are retrieved into the vAGC are cached by the cache server 310 . They are requested by the AGS through the cache server 310 . If a cached copy of the requested data exists in the cache server, it is delivered directly to the AGS. If not, the cache server retrieves the data, caches it and delivers the data to the AGS to fulfill the request.
- the AGS in order to simplify the design of the AGS and to improve the performance and scalability of it, the AGS is designed to handle only one native media format.
- one suitable format for audio is G.711 or GSM.
- Media that come in different format are handed over to the media conversion proxy server 320 , which coverts the media to the native format of the AGS 200 .
- FIG. 7 is a block diagram illustrating schematically the components of the media conversion proxy server.
- the media conversion proxy server comprises a text-to-speech module 322 , a speech-to-text module 324 , an audio conversion module 326 and a protocol conversion module 328 .
- the modular design allows for other “plug-ins” as the need arises.
- the text-to-speech module 322 is used for converting text to synthesized speech. For example, this is useful for reading back e-mail messages.
- the speech-to-text module 324 is used for converting speech to text. This is useful in speech recognition applications involving responding to a user's voice response.
- the audio conversion module 326 converts between a supported set of audio formats, such as G.711, G.723, CD audio, MP3, . . . .
- the protocol conversion module 328 allows conversions between protocols such as IMAP (Internet Message Access Protocol) and SMTP (Simple Mail Transfer Protocol).
- FIG. 8 is a detailed block diagram of the Application Gateway Server, which is the main component of the Application Gateway Center.
- the Application Gateway Server (AGS) 200 is responsible for accepting incoming calls, retrieving the vAPP associated with the dialed number and executing the vXML scripts of the vAPP. Each incoming call is treated as a separate session and the AGS is responsible for processing all user events and system actions that occur in multiple simultaneous sessions. The AGS is also responsible for all call routing in all sessions.
- the AGS 200 is a set software modules running on a Windows NT or UNIX server.
- the AGS is implemented as a Windows NT machine on a card, and multiple cards are installed on a caged backplane to form a high scalable system.
- the AGS 200 comprises four main software modules, a session manager 210 , an I/O abstraction layer 220 , a computer telephony (CT) abstraction layer 230 , and a telephony scripting language parser 240 .
- the telephony scripting language parser 240 further comprises a telephony XML or vXML parser 242 and a generic XML parser 244 .
- a streaming interface 250 provides a direct streaming path for media data between the I/O abstraction layer 220 and the CT abstraction layer.
- Each of these modules is designed to be a separate DLL (Dynamically Linked Library) and perform a specific task.
- the AGS is a console only application with no user interface for any of these modules.
- Several of these modules incorporate commercial, third party software components in performing their tasks. These components will be discussed along with the appropriate modules.
- the session manager 210 is the centerpiece of the AGS 200 . It is responsible for creating new sessions, deleting terminated sessions, routing all actions and events to the appropriate modules and maintaining modularity between each session. It responds to I/O and vXML goto requests, and other additional events. In one embodiment, it employs commercially available software libraries containing thread and string classes from PWLib, a product of Equivalence Pty Ltd, Erina, New South Wales, Australia.
- the session manager interfaces to the external of the AGS via the I/O abstraction layer 220 and the CT abstraction layer 230 . It accesses the I/O and CT layers as a set of classes and member functions that are individual DLLs.
- the Session Manager 210 runs as a single-threaded processor of actions and event.
- FIG. 8 also illustrates the manner in which the modules of the AGS must communicate with each other.
- the session manager communicates to both the I/O abstraction layer and the CT abstraction layer through traditional DLL entry points with C/C++ parameter passing.
- the I/O abstraction layer and the CT abstraction layer communicate through a streaming interface.
- the session manager and the telephony scripting language parser communicate through DLL entry points using microXML.
- the session manager 210 behaves like a virtual machine with its own set of “OpCodes”. MicroXML is the parsed vXML scripts interpreted into these OpCodes, and will be described in more detail later.
- a session begins with the reception of an asynchronous event from the CT abstraction module 230 signaling an incoming call.
- the Session Manager then creates a session for this call by accessing a database (e.g. DIR 1 of FIG. 4A ) keyed on the session's DNS and ANI information, which returns an initial vXML script.
- the telephony scripting language parser 240 is a separate DLL invoked through short microXML event scripts. It returns a microXML action script. A cycle of actions and events begins with the transmission of this script to the telephony scripting language parser 240 for processing.
- the telephony scripting language parser 240 responds to this event by returning a simple vXML script of its own containing I/O and CT action requests collected from the parsing of the script.
- the Session Manager now processes these action requests and then returns to parsing until the end of the session.
- SID session ID
- GUID globally unique Ids
- the session manager 210 is accessed or invoked via a number of interface points of its DLL as described in TABLE 1.
- VXESessionManager( ) VXESessionManager constructor function. It creates and starts up an instance of an AGS Session Manager. ⁇ VXESessionManager( ) VXESessionManager destructor function. It shuts down and deletes an instance of an AGS Session Manager. AddEvent(VXEEvent&) Member function to submit an event to a Session Manager's event queue. It receives a record of the incoming event and outputs TRUE if submission is successful, FALSE otherwise. GetSessions( ) Provides a count of active sessions.
- the I/O abstraction layer 220 performs all input and output operations for the AGS 200 . Essentially, it renders transparent to the internal of the AGS the variety of I/O formats and protocols that might be encounter externally.
- To the session manager 210 most HTTP, FTP, File, and memory-mapped I/O requests are reduced to four commands: open, close, read, and write. This allows access to a stream from any of these sources with the same procedure calls once the stream is open.
- it incorporates available commercial software libraries, such as WinInet from Microsoft Corporation, Seattle, Wash., U.S.A and PWLib from Equivalence Pty Ltd. WinInet is a windows-specific DLL that allows the I/O abstraction layer to communicate to outside sources using HTTP and FTP.
- PWLib also used by the session manager 210 contains strings and threads classes.
- the I/O abstraction layer 220 is accessed or invoked via a number of interface points of its DLL as described in TABLE 2.
- a single thread per active stream is created by instantiating a VXEIOStream when accessed by the session manager 210 . If the stream is FTP or HTTP-based, then the user will need to provide the appropriate login data, submission method, and CGI variables. Next, the user calls the Open method and then uses the Read and Write methods to operate upon the stream until closing it with the Close method. At this point, this instance of the VXEIOStream is available for use on another stream source or it can be deleted.
- VXEIOStream( ) VXEIOStream constructor function It creates a new instance of a VXEIOStream. ⁇ VXEIOStream( ) VXEIOStream destructor function. It shuts down stream and releases associated memory open/openAsynchronous(char* Member function to open an I/O stream either name, StreamType streamtype, synchronously or asynchronously.
- OpenMode mode pathname
- type of stream HTTP, FTP, File, or Memory
- Opening mode Read/Write
- TRUE/FALSE success/failure in synchronous mode, corresponding event asynchronously.
- close( ) Member function to close an open stream. It outputs: TRUE/FALSE for success/failure.
- read/readAsynchronous(void* Member function to read from an open stream either buffer, int count) synchronously or asynchronously. It has inputs: Pointer to buffer into which to write data, byte count to read from stream.
- AddCGIVariable VXEVariable& Member function to add a CGI variable for v) submission to an HTTP stream before opening it. It has inputs: Variable name/value pair contained in a VXEVariable class. It has outputs: TRUE/FALSE for success/failure.
- SetFTPLogin PString& name, Member function to set FTP login information for an Pstring& passwd) FTP stream before opening it. It has inputs: FTP user name and password.
- the computer telephony (CT) abstraction layer 230 is a thin abstraction layer that makes it possible for the AGS 200 to communicate with several computer telephony devices and/or protocols.
- the CT abstraction layer receives requests for computer telephony actions from the session manager 210 and translates those requests to a CT module.
- the CT abstraction layer receives user events directed to that CT module and relates them back to the session manager.
- the CT modules include a H.232 stack for handling VoIP signals, a SIP (Session Interface Protocol), a MGCP (Media Gateway Control Protocol) as well as other CT modules such as Dialogic CT modules. Since several CT modules can be placed below the CT abstraction layer and the CT abstraction will talk to all of the CT modules, the modular design allows the AGS to communicate with a new computer telephony device or protocol simply with the addition of a new CT module.
- the CT abstraction layer 230 will preferably make use of PWLib's platform-independent thread class.
- the CT Abstraction layer is instantiated by the Session Manager 210 . It then seeks out a vXML configuration file that contains information on the number and type of telephony boards in its system.
- the member functions represent generic functionality that should be supportable across a wide variety of telephony hardware.
- the motivation for this abstraction layer is to make the AGS 200 both platform and protocol independent.
- the Session Manager 210 the Session Manager 210 , XML Parser 240 , and CT Abstraction layer 230 cooperate via the following protocol.
- the telephony scripting language parser 240 locates a vXML element which requires a telephony task.
- the telephony scripting language parser sends this task to the Session Manager in a microXML action string.
- the Session Manager parses the microXML action string and determines the appropriate call to the CT abstraction layer along with its associated parameters.
- the Session Manager now calls the CT abstraction layer asynchronously and the CT abstraction layer returns an event signaling the completion of the CT task and the Session Manager resumes parsing.
- the CT abstraction layer 230 is accessed or invoked via a number of interface points of its DLL as described in TABLE 3.
- VXECTAbstraction (VXESessionManager*) VXECTAbstraction constructor function. It has input: Associated Session Manager. It creates a new instance of a CT Abstraction. ⁇ VXECTAbstraction( ) VXECTAbstraction destructor function. It shuts down an instance of a Voxeo CT Abstraction and releases associated memory GetVersion(PString& version) Member function to determine version. It has inputs: Reference to a string into which to copy version information. It has outputs: Version information copied into parameter 1 string GetProtocol(PString& protocol) Member function to determine active telephony protocol.
- VXECall* inputs: Associated session, number to call. It has outputs: Asynchronous event indicating success/failure sent to Session Manager. dial(VXESession* pSession, Member function to dial a string of digits. It has Pstring* number) inputs: Associated session, digits to dial. It has outputs: Asynchronous event indicating success/failure sent to Session Manager. Wink(VXESession* pSession) Member function to perform wink function. It has inputs: Associated session. It has outputs: Asynchronous event indicating success/failure sent to Session Manager to an HTTP stream before opening it.
- Void conference (VXESession* Member function to conference two active pSession1, VXESession* sessions/calls. It has inputs: Two sessions to pSession2) conference together. It has outputs: Asynchronous event indicating success/failure sent to Session Manager. Void Member function to flush digit buffer. It has flushDigitBuffer(VXESession* inputs: Associated session. It has outputs: pSession) Asynchronous event indicating success/failure sent to Session Manager. Void getDigits(VXESession* Member function to read digits from digit buffer.
- const PString& date const PString& inputs: Associated session, string containing termdigits) desired date, termination digits string. It has outputs: Asynchronous event indicating completion/termination sent to Session Manager. PlayTime(VXESession* pSession, Member function to play current time. It has const PString& time, const inputs: Associated session, string containing PString& termdigits) desired time, termination digits string. It has outputs: Asynchronous event indicating completion/termination sent to Session Manager. PlayMoney(VXESession* pSession, Member function to play a dollar value.
- const float value const PString& inputs: Associated session, value to play, termdigits) termination digits string. It has outputs: Asynchronous event indicating completion/termination sent to Session Manager.
- PlayCharacters(VXESession* Member function to play a string of characters. It pSession, const PString& string, has inputs: Associated session, string of characters const Pstring& termdigits) to play, termination digits. It has outputs: Asynchronous event indicating completion/termination sent to Session Manager.
- PlayString(VXESession* pSession, Member function to pronounce a text message.
- const PString& string const has inputs: Associated session, string to Pstring& termdigits) pronounce, termination digits. It has outputs: Asynchronous event indicating completion/termination sent to Session Manager.
- PlayNumber(VXESession* Member function to play a number. It has inputs: pSession, const PString& number, Associated session, string containing number to const Pstring&termdigits) pronounce, termination digits. It has outputs: Asynchronous event indicating completion/termination sent to Session Manager.
- PlayOrdinal(VXESession* Member function to play an ordinal (1st, 2nd, 2rd, pSession, const PString& ordinal, etc.).
- ReceiveFAX(VXESession* Member function to receive a FAX. It has inputs: pSession, VXEIOStream &file) Associated session, VXEIOStream to which to receive FAX. It has outputs: Asynchronous event indicating success/failure sent to Session Manager.
- the streaming interface 222 provides a direct streaming transfer between the I/O abstraction layer 220 and the CT abstraction layer 230 when media data, such as audio or other multimedia is involved.
- the streaming interface facilitates the AGS to play audio from URL's and to record audio to URL's in a streaming manner.
- the interface is generic and passes the burden of buffer management to the CT module in use. This allows specific CT modules to buffer information as appropriate for the corresponding telephony hardware or protocol.
- the streaming interface is implemented through the readAsynchronous and writeAsynchronous interface points in the I/O abstraction layer.
- the telephony scripting language parser 240 is responsible for parsing the vXML scripts handed to it by the session manger 210 . It in turn informs the session manager of the described actions coded in the vXML scripts.
- the telephony scripting language parser is modular and can accommodate additional parsers such as that for voiceXML and parsers for other telephony scripting language that may arise. In the present preferred embodiment, it comprises the vXML parser 242 and the generic XML parser 244 .
- the generic XML parser 244 parses the vXML scripts, which are essentially XML scripts with embedded custom telephony tags, and puts them in a format that the vXML parser 242 can expediently act on.
- the generic XML parser 244 conveniently employs CueXML components available from CueSoft, Inc, Brighton, Colo., U.S.A. These components enable parsing of vXML documents into an object model, DOM (Document Object Model) listing the parsed objects in a hierarchical tree structure. This allows the vXML parser 242 , which in the preferred embodiment is a DLL written in Delphi 5.0, to “walk” through the tree of objects and interpret them into microXML codes that can be understood by the session manager 210 .
- the vXML parser 242 behaves as follows: when called it will examine the incoming microXML and determine if there is a buffer of new vXML to parse, if such a buffer exists then the parser uses the generic XML parser 244 to construct a new object model for this buffer, the session object model is set to that model and the session state is cleared.
- the vXML parser 242 begins parsing from the session state in the session object model (an empty state implies the beginning of a document). As the parse traverses the document model the state is updated and events are generated. If these events are internal to the processor they are handled (i.e.
- the vXML parser 242 is required to maintain state per session so that each invocation of the vXML parser will continue where the previous invocation within the same session ended.
- the maintenance of state includes preserving the DOM for the current instance of vXML, the node in the DOM that the parser is currently examining, and any variables that are associated with the session.
- the vXML parser 242 is accessed or invoked via a number of interface points of its DLL as described in TABLE 4.
- VXML PARSER Interface Points DESCRIPTION Create Creates an instance of the vXML parser. It has output: integer result code (negative numbers denote errors). Destroy Destroys an instance of the vXML parser. It has output: integer result code (negative numbers denote errors). Parse Performs the main tasks of the vXML parser (i.e. determines actions from vXML, and maintains state. It inputs: microXML string containing the sessionID. The microXML may also contain a buffer of vXML (described above) and a pointer to instance data. It outputs: microXML string containing the action(s) generated by this invocation and possibly modification of the instance data. Kill It has input: pointer to instance data. It has output: integer result code (negative numbers denote errors).
- microXML is a subset of simple vXML used for communication between the session manager 210 and the telephony scripting language parser 240 .
- MicroXML is the native codes of the virtual machine of the session manager 210 .
- the vXML parser 242 communicates with the session manger 210 in a synchronous manner using microXML.
- user events may also be reported to the vXML parser via microXML. If a user event is reported the parser will find the appropriate event handler by first looking locally for a valid handler. If a handler is not found there then the parent node in the document model is examined for a valid handler.
- MicroXML is composed of a limited number of tags, these tags do not have any attributes, and CDATA sections are not supported.
- Table 5 shows examples of microXML tags:
- vXML is XML with additional custom tags for telephony applications.
- TABLE 6A-6D lists example tags useful for creating telephony applications. A user or developer need only code his or her telephony application in these vXML tags and deploy the resulting scripts as a webpage on the Internet for the vAGS 200 to access.
- method “put
- One example of Return is to implement ⁇ return/> ⁇ goto> calls as a call stack.
- #block” or URI in a new session, then continues to submit “all
- x,y,z” process this session method “put
- FALSE” of making a function to gather digits from includetermdigit “TRUE
- the audio file will be played 3 times ⁇ onhangup> ⁇ /onhangup> before “timing out” and moving on in the ⁇ /menu> vXML code.
- ⁇ inputdigits label “input_pin”
- FALSE” of making a function to gather digits from includetermdigit “TRUE
- the following are examples of microXML communication between the session manager 210 and the vXML parser 242 .
- the example vXML file results in the following corresponding microXML being generated by the vXML parser and sent to the session manager:
- FIG. 9 is a system block diagram of a network traffic monitoring system operating in cooperation with the Distributed Application Telephony Network System of the present invention.
- the invention contemplates a larger number of enterprises and users will deploy telephony applications on the Internet 30 in the form of vXML applications such as vAPP 1 , vAPP 2 , . . . , vAPP m. These applications are served by a number of web servers 46 on the Internet. When a call associated with one of the these vAPP enters the Internet, it must be directed to one of a pluarality of application gateway centers, such as vAGC 1 , vAGC 2 , . . . , vAGC n.
- the call in order to provide the best quality-of-service, is preferably directed to a vAGC having the best access to the associated vAPP.
- the invention includes providing monitoring of the accessibility of the individual vAPPs relative to the plurality of vAGCs. This will enable the call to be directed to the vAGC having the best access to that associated vAPP.
- Each vAGC site is provided with a traffic monitor 400 that periodically pings the pluarality of vAPP sites and detects the return signals.
- the response time of each vAPP site to any given vAGC is collected by a network monitoring server 410 . Since each vAPP is associated with a dialed number (DN), the network monitoring server computes a listing of DNs versus vAGCs sorted in order of fastest response time. This information is used to update the DIR 0 directory (see FIG. 4A ) dynamically. In this way, when a call to a given DN is to be directed to an AGC, a DIR 0 lookup will point to the vAGC with the faster access to the vAPP site associated with the given DN.
- DN dialed number
- FIG. 10 illustrates a preferred network configuration including the PSTN and the Internet for practicing the invention.
- the network is similar to that shown in FIG. 4A or FIG. 5 where the PSTN 10 is a network of telephones connectable by switched circuits and the Internet 30 is a network of IP devices and resources communicating by IP packets.
- a set of voice applications scripted in vXML 110 - 1 to 110 - m is hosted by corresponding web servers 112 - 1 to 112 - m and is accessible on the Internet.
- a plurality of voice application gateway centers (“vAGC’) 100 (also referred to as “voice centers”) is deployed on the Internet.
- each vAGC 100 essentially servers as a “browser” for one of the vXML voice applications and processes a received call by executing the vXML script.
- These vAGCs or voice centers can be divided into at least two categories depending on whether they are maintained and operated in hosted facilities 500 or on a subscriber's premise 510 .
- vAGCs 100 - 1 to 100 - n in one or more hosted facilities 500 are installed and operated by a telephony service provider.
- a customer or subscriber may purchase or lease a vAGC such as vAGC P 1 and operate the voice center on premise 1 .
- vAGC P 1 may be purchased or lease a vAGC
- vAGC P 2 located on premise 2 .
- the various vAGCs are set up to received a call and process it according to a vXML voice application associated with the call's dialed number DN.
- One or more access servers 14 route calls between the PSTN and the Internet. When a call is directed to the Internet, the access server 14 looks up a destination vAGC in a voice-center directory, DIR 0 , and routes the call there. Once the vAGC has received the call, it looks up DIR 1 for the URL of the vXML application associated with the dialed number. The vAGC then retrieves and executes the vXML script to process the call.
- the access server 14 routes the call to a vAGC routing proxy 580 which then takes over the job of the access server and routes the call to a destination vAGC after a DIR 0 lookup.
- a vAGC routing proxy 580 which then takes over the job of the access server and routes the call to a destination vAGC after a DIR 0 lookup.
- a network system and method allows sharing of resources between voice application gateway centers (“voice centers”) operated by a hosted facility and by a subscriber or customer on premise.
- voice centers voice application gateway centers
- a customer is able to configure the network system to have some calls processed on premise and some calls processed by the host facility.
- a hosted facility is able to configure the network system to have some calls originally to be processed by the hosted facility to be processed by an available premise voice center.
- the invention is accomplished by a semi-real-time manipulation of a voice-center directory by which a destination voice center is selected to process a call.
- At least a provisioning management server with a user interface for a user to easily update lookup information in the voice-center directory.
- the lookup information enables a voice center to be selected as a function of the dialed number as well as routing rules.
- the user either an operator from the hosted facility or a customer with premise equipment, is able to update the routing rules in the voice-center directory.
- FIG. 10 also shows one or more provisioning management server 600 maintained by the hosted facility 500 .
- the provisioning management server 600 has a user interface and enables a user to update directory entries in the voice-center directory, DIR 0 550 as well as the voice-application directory, DIR 1 570 .
- the user interface is a standard one such as a web interface that can be easily accessed by users 610 -H from the hosted facility 500 as well as users 610 -P 1 , . . . , 610 -Pm on premise.
- FIG. 11 is a block diagram of the provisioning management server.
- the server 600 comprises a manager 610 for managing the operations, an event monitor 620 , a user interface 630 , a directory replicator 650 , a file storage 640 and an I/O and network interface 660 .
- the file storage 640 stores replica 642 of provisioning data from DIR 0 550 and a set of predefined provisioning and routing rules 644 . It also stores replica 646 of addresses of voice applications from DIR 1 570 and customer accounts 648 .
- the manager 610 manages user access and responds to the event monitor 620 to initiate various operations.
- a user can add modify or delete data in the replica 642 and 646 .
- the data pertaining to provisioning and routing rules may be selected from the predefined set 644 .
- the directory replicator replicates relevant portion of the data between the replica 642 and DIR 0 550 as well as between the replica 646 and DIR 1 570 so that changes made to the replica in the provisioning management server are reflected in DIR 0 550 and DIR 1 570 .
- provisioning management server 600 a user including a subscriber can quickly make changes to DIR 0 550 to implement close to real-time re-allocation of resources and services between premise and hosted facilities
- FIG. 12 illustrates the information contained in an entry in the voice-center directory.
- the allocation of resources among the vAGC or voice-centers of the hosted facility and on a customer's premise is configurable by modifying the directory entries in the voice-center directory, DIR 0 550 .
- a lookup of DIR 0 yields a prioritized list of vAGCs or voice-centers for processing the current call.
- the order of listed voice centers is a function of the dialed number DN, the routing rules associated with the DN and the capacity or availability of the voice centers.
- the routing rules is able to implement weighting and priority rules for routing calls between vAGC's that are either hosted or at customer premise sites. For example, referring to FIG.
- one such routing rules requires 50% of the call traffic to be routed to vAGC 1 on hosted site and the other 50% to vAGC P 1 on the customer premise site. If both vAGC 1 and vAGC P 1 are unavailable, the call will be routed to vAGC 3 on the hosted site.
- the voice-center lookup directory DIR 0 550 essentially functions as a directory for looking up call-processing resources analogous to a domain name server (“DNS”) for locating the IP address of an Internet resource given an URL address.
- DIR 0 is an ENUM directory.
- ENUM is the name of a protocol that resolves fully qualified telephone numbers to fully qualified domain name addresses using a DNS-based architecture. It is the name of a chartered working group of the Internet Engineering Task Force (IETF) chartered to develop protocols that map telephone numbers to resources found on the Internet using the Domain Name System.
- the voice-center selection function of the voice-center directory also depends on the degree of availability or current capacity of individual voice centers. This is accomplished by having the individual voice centers updating the voice-center directory regarding their current capacity at predetermined time intervals.
- the vAGCs 100 or voice centers are configured to update DIR 0 as to their availability or their current capacity at predetermined time intervals.
- the availability information from individual voice centers is first sent to the vAGC routing proxy 560 to be cached there before being forwarded to update DIR 0 .
- the lookup will return a prioritized list of vAGC's.
- the access server 14 or the routing proxy 560 will then pick the vAGC from the top of the list and attempt to route the call there.
- FIG. 13 illustrates another example of a voice center updating its current capacity or availability information.
- vAGC 1 sends information about it availability every 30 seconds to both the voice-center directory DIR 0 550 and to the vAGC routing proxy 560 .
- FIG. 14 illustrates an example of user configurable call routing between voice centers located on different sites.
- DIR 0 550 is consulted to locate the list of prioritized vAGC's that can service the call.
- the call is directed to vAGC P 1 on a customer's premise site. If for some reason vAGC P 1 failed (e.g., had no more capacity or was offline due to network problems) the call will be routed to the next vAGC down the list. (3) In this case, it is re-routed to vAGC 1 in a hosted site.
- FIG. 15 illustrates another example of user configurable call routing using the vAGC routing proxy instead of the access server.
- the call is passed to a vAGC routing proxy 560 .
- the vAGC routing proxy performs a DIR 0 lookup to obtain a prioritized list of vAGC's that can service the call.
- the call is directed to vAGC P 1 on a customer's premise site. If for some reason vAGC P 1 failed (e.g., had no more capacity or was offline due to network problems) the call will be re-routed by the routing proxy 560 to the next vAGC down the list. (4) In this case, it is routed to vAGC 1 in a hosted site.
- the network of telephony system includes a voice center receiving a routed call and retrieving a voice application associated with the dialed number and appropriate for that voice center.
- the call can potentially be routed to any one of a number of voice centers, preferably there is a version of the voice application appropriate for each of the number of voice centers.
- a local network address of the appropriate version of the voice application is cache in each voice center so that the voice center can use it to retrieve the voice application without having to perform a look up in an external directory.
- FIG. 16 illustrates maintaining different versions of voice applications for a given dialed number.
- a voice application designed for a customer site may no longer be appropriate if the call is routed to a hosted site.
- the interactive voice response (“IVR”) may involve a human attendant on a customer site but not so in a hosted site.
- IVR interactive voice response
- Another example is that different sites may have different firewall settings and even the same version may need to be hosted at different URLs for accessibility.
- IVR interactive voice response
- different versions of voice application for a given dialed number are maintained so that when a call is routed to a voice center of a certain location or environment, an appropriate version is used to process the call.
- vAGC P 1 the URL 574 ′ of the version of the vAPP appropriate for that site.
- the premise vAGC P 1 maintains a local replica of the premise version 574 ′′ of DIR 1 data by a replication agent 552 .
- a premise-based vAGC can operate even when the link to the hosted DIR 1 is severed.
- FIG. 17 is a flow diagram illustration the scheme for allocation of voice service and resource in a hybrid environment of premise and hosted sites.
- FIGS. 18A-18J illustrate embodiments of the routing rules of STEP 740 in FIG. 17 .
- FIG. 18A illustrates one embodiment of the routing rules of STEP 740 in FIG. 17 .
- STEP 742 Wherein the user is a subscriber, and routing rules configurable by the subscriber include telephone calls to be preferentially processed by an application gateway center on premise from the second set and to be processed by a hosted application gateway center from the first set in the event of the operating capacity of the second set fall below a predetermined threshold.
- FIG. 18B illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- STEP 744 Wherein the user is a host operator, and routing rules configurable by the host operator include telephone calls to be processed by an application gateway center from the second set in the event of the operating capacity of the first set falls below a predetermined threshold.
- FIG. 18C illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- STEP 746 Wherein the user is a subscriber, and routing rules configurable by the subscriber include telephone calls each having one of a predefined dialed numbers must be processed by an application gateway center on premise from the second set for security reasons.
- FIG. 18D illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- STEP 748 Wherein the user is a subscriber, and routing rules configurable by the subscriber include telephone calls to be preferentially processed by a hosted application gateway center from the first set when the telephone calls are associated with a first predefined set of telephone numbers.
- FIG. 18E illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- routing rules configurable by the user include routing a predetermined percentage of call traffic to application gateway centers on premise from the second set relative those from the first set.
- FIG. 18F illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- routing rules configurable by the user include basing on geographical location of the call to be processed.
- FIG. 18G illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- routing rules configurable by the user include basing on time schedule.
- FIG. 18H illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- routing rules configurable by the user include basing on the carrier the call originates from.
- FIG. 18I illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- routing rules configurable by the user include avoidance of predefined network routes.
- FIG. 18J illustrates another embodiment of the routing rules of STEP 740 in FIG. 17 .
- STEP 759 Wherein the user is a subscriber, and routing rules configurable by the subscriber include routing from one premise to another.
- the hybrid resource allocation system described also offers the following business advantages. It allows a service provider to sell premise-based telephony software and hosted telephony services as a combined product/service, such that the customer's premise telephony software ports integrate seamlessly with the service provider's hosted telephony software ports.
- the customer will be allowed to use an online management console to configure and manage his premise-based software, making it possible for him to set parameters for how many calls he wants the premise system (as opposed to the hosted system) to take, under what circumstances to route calls through the hosted or premise system.
- the customer does not have to guess what his traffic will be. He can buy 100 ports of software, and simply use the hosted network to take all calls after the first 100 concurrent calls. Or he can buy 20 ports and take only the calls about a certain subject on premise, with the rest going through the hosted service. Or he can take calls on-premise during business hours, and run them through hosted at night. Whatever the customer wants to do, he now has the flexibility of the premise and hybrid models.
Abstract
Description
TABLE 1 |
Session Manager's Interface Points |
SESSION MANAGER | |
Interface Points | DESCRIPTION |
VXESessionManager( ) | VXESessionManager constructor function. |
It creates and starts up an instance of an AGS | |
Session Manager. | |
~VXESessionManager( ) | VXESessionManager destructor function. |
It shuts down and deletes an instance of an | |
AGS Session Manager. | |
AddEvent(VXEEvent&) | Member function to submit an event to a |
Session Manager's event queue. It receives a | |
record of the incoming event and outputs | |
TRUE if submission is successful, FALSE | |
otherwise. | |
GetSessions( ) | Provides a count of active sessions. |
TABLE 2 |
I/O Abstraction Layer's Interface Points |
I/O ABSTRACTION LAYER | |
Interface Points | DESCRIPTION |
VXEIOStream( ) | VXEIOStream constructor function. It creates a |
new instance of a VXEIOStream. | |
~VXEIOStream( ) | VXEIOStream destructor function. It shuts down |
stream and releases associated memory | |
open/openAsynchronous(char* | Member function to open an I/O stream either |
name, StreamType streamtype, | synchronously or asynchronously. It has inputs: |
OpenMode mode) | pathname, type of stream (HTTP, FTP, File, or |
Memory), and opening mode (Read/Write); and | |
output: TRUE/FALSE for success/failure in | |
synchronous mode, corresponding event | |
asynchronously. | |
close( ) | Member function to close an open stream. It |
outputs: TRUE/FALSE for success/failure. | |
read/readAsynchronous(void* | Member function to read from an open stream either |
buffer, int count) | synchronously or asynchronously. It has inputs: |
Pointer to buffer into which to write data, byte count | |
to read from stream. It has outputs: Number of | |
bytes read synchronously, corresponding event | |
asynchronously | |
write/writeAsynchronous(void* | Member function to write to an open stream either |
buffer, int count) | synchronously or asynchronously. It has inputs: |
Pointer to buffer from which to write data, byte | |
count to write to stream. It has outputs: Number of | |
bytes written synchronously, corresponding event | |
asynchronously. | |
GetPos( ) | Member function to return position within a stream. |
SetSubmitMethod(SubmitMethod | Member function to set CGI submission method for |
method) | an HTTP stream before opening it. It has inputs: |
Submission method, either GET or PUT. | |
AddCGIVariable (VXEVariable& | Member function to add a CGI variable for |
v) | submission to an HTTP stream before opening it. It |
has inputs: Variable name/value pair contained in a | |
VXEVariable class. It has outputs: TRUE/FALSE | |
for success/failure. | |
SetFTPLogin(PString& name, | Member function to set FTP login information for an |
Pstring& passwd) | FTP stream before opening it. It has inputs: FTP |
user name and password. | |
TABLE 3 |
CT Abstraction Layer's Interface Points |
CT ABSTRACTION LAYER | |
Interface Points | DESCRIPTION |
VXECTAbstraction(VXESessionManager*) | VXECTAbstraction constructor function. It has |
input: Associated Session Manager. It creates a | |
new instance of a CT Abstraction. | |
~VXECTAbstraction( ) | VXECTAbstraction destructor function. It shuts |
down an instance of a Voxeo CT Abstraction and | |
releases associated memory | |
GetVersion(PString& version) | Member function to determine version. It has |
inputs: Reference to a string into which to copy | |
version information. It has outputs: Version | |
information copied into |
|
GetProtocol(PString& protocol) | Member function to determine active telephony |
protocol. It has inputs: Reference to a string into | |
which to copy protocol information. It has | |
outputs: Protocol information copied into | |
|
|
Answer(VXESession* pSession) | Member function to answer an incoming call. It |
has inputs: Session associated with incoming call. | |
It has outputs: Asynchronous event indicating | |
success/failure sent to Session Manager. | |
Hangup(VXESession* pSession) | Member function to hang up on an active call. It |
has inputs: Session associated with active call. It | |
has outputs: Asynchronous event indicating | |
success/failure sent to Session Manager. | |
call(VXESession* pSession, | Member function to make an outgoing call. It has |
VXECall*) | inputs: Associated session, number to call. It has |
outputs: Asynchronous event indicating | |
success/failure sent to Session Manager. | |
dial(VXESession* pSession, | Member function to dial a string of digits. It has |
Pstring* number) | inputs: Associated session, digits to dial. It has |
outputs: Asynchronous event indicating | |
success/failure sent to Session Manager. | |
Wink(VXESession* pSession) | Member function to perform wink function. It has |
inputs: Associated session. It has outputs: | |
Asynchronous event indicating success/failure sent | |
to Session Manager to an HTTP stream before | |
opening it. | |
Void conference(VXESession* | Member function to conference two active |
pSession1, VXESession* | sessions/calls. It has inputs: Two sessions to |
pSession2) | conference together. It has outputs: |
Asynchronous event indicating success/failure sent | |
to Session Manager. | |
Void | Member function to flush digit buffer. It has |
flushDigitBuffer(VXESession* | inputs: Associated session. It has outputs: |
pSession) | Asynchronous event indicating success/failure sent |
to Session Manager. | |
Void getDigits(VXESession* | Member function to read digits from digit buffer. |
pSession, int maxdigits, Pstring& | It has inputs: Associated session, maximum digits |
termdigits, Pstring& outdigits) | to read, termination digits string, string for digits |
read. It has outputs: Asynchronous event | |
indicating success/failure and digits read sent to | |
Session Manager. | |
PlayStream(VXESession* | Member function to play audio from an open |
pSession,VXEIOStream&, const | stream. It has inputs: Associated session, audio |
Pstring& termdigits) | stream, and termination digits. It has outputs: |
Asynchronous event indicating | |
completion/termination sent to Session Manager. | |
PlayDate(VXESession* pSession, | Member function to play current date. It has |
const PString& date, const PString& | inputs: Associated session, string containing |
termdigits) | desired date, termination digits string. It has |
outputs: Asynchronous event indicating | |
completion/termination sent to Session Manager. | |
PlayTime(VXESession* pSession, | Member function to play current time. It has |
const PString& time, const | inputs: Associated session, string containing |
PString& termdigits) | desired time, termination digits string. It has |
outputs: Asynchronous event indicating | |
completion/termination sent to Session Manager. | |
PlayMoney(VXESession* pSession, | Member function to play a dollar value. It has |
const float value, const PString& | inputs: Associated session, value to play, |
termdigits) | termination digits string. It has outputs: |
Asynchronous event indicating | |
completion/termination sent to Session Manager. | |
PlayCharacters(VXESession* | Member function to play a string of characters. It |
pSession, const PString& string, | has inputs: Associated session, string of characters |
const Pstring& termdigits) | to play, termination digits. It has outputs: |
Asynchronous event indicating | |
completion/termination sent to Session Manager. | |
PlayString(VXESession* pSession, | Member function to pronounce a text message. It |
const PString& string, const | has inputs: Associated session, string to |
Pstring& termdigits) | pronounce, termination digits. It has outputs: |
Asynchronous event indicating | |
completion/termination sent to Session Manager. | |
PlayNumber(VXESession* | Member function to play a number. It has inputs: |
pSession, const PString& number, | Associated session, string containing number to |
const Pstring&termdigits) | pronounce, termination digits. It has outputs: |
Asynchronous event indicating | |
completion/termination sent to Session Manager. | |
PlayOrdinal(VXESession* | Member function to play an ordinal (1st, 2nd, 2rd, |
pSession, const PString& ordinal, | etc.). It has inputs: Associated session, ordinal, |
const Pstring& termdigits) | termination digits. It has outputs: Asynchronous |
event indicating completion/termination sent to | |
Session Manager. | |
RecordStream(VXESession* | Member function record to an open |
pSession, XEIOStream& stream) | VXEIOStream. It has inputs: Associated session, |
target stream. It has outputs: Asynchronous event | |
indicating success/failure sent to Session Manager. | |
SendFAX(VXESession* pSession, | Member function to send a FAX. It has inputs: |
VXEIOStream& file) | Associated session, VXEIOStream containing data |
to FAX. It has outputs: Asynchronous event | |
indicating success/failure sent to Session Manager. | |
ReceiveFAX(VXESession* | Member function to receive a FAX. It has inputs: |
pSession, VXEIOStream &file) | Associated session, VXEIOStream to which to |
receive FAX. It has outputs: Asynchronous event | |
indicating success/failure sent to Session Manager. | |
TABLE 4 |
vXML Parser Interface Points |
VXML | |
PARSER | |
Interface | |
Points | DESCRIPTION |
Create | Creates an instance of the vXML parser. It has output: |
integer result code (negative numbers denote errors). | |
Destroy | Destroys an instance of the vXML parser. It has output: |
integer result code (negative numbers denote errors). | |
Parse | Performs the main tasks of the vXML parser (i.e. |
determines actions from vXML, and maintains state. It | |
inputs: microXML string containing the sessionID. The | |
microXML may also contain a buffer of vXML (described | |
above) and a pointer to instance data. It outputs: | |
microXML string containing the action(s) generated by | |
this invocation and possibly modification of the instance | |
data. | |
Kill | It has input: pointer to instance data. It has output: |
integer result code (negative numbers denote errors). | |
TABLE 5 |
microXML Tags |
MicroXML TAG | NAME | MicroXML TAG | NAME |
ACT | Action | EVL | Event Value |
BUF | Buffer | LBL | Label |
DAT | Instance Data | TYP | Type |
ERR | Error | P00 | Parameter0 |
EVT | Event | P00 | Parameter99 |
ETP | Event Type | SID | Session ID |
TABLE 6A |
vXML General Tags |
vXML GENERAL TAG | |
Examples | DESCRIPTION |
<assign var=“ttt” | Assigns value “123” to variable named “ttt. |
value=“123”/> | |
<clear var=“ttt”/> | Clears variable named “ttt”. |
<clearDigits /> | Clears the digit buffer. |
<getDigits | This element reads input digits from the |
var=“pager_msg” | phone and places them into a variable |
maxdigits=“9” | defined within the element itself. In the |
termdigits=“#*” | example, the user would have 30 seconds to |
includetermdigit=“TRUE|FALSE” | enter up to 9 digits on her phone, pausing |
cleardigits=“TRUE|FALSE” | no more than 5 seconds between digits, and |
maxtime=“30s” | ending the digit input with either the # key |
ptime=“5s”/> | or * key. This element is designed for |
gathering PIN codes, Pager numbers, and | |
anything else that involves multiple digits | |
coming from the user. | |
<block | The block element is used to logically |
label=“here” | group other elements together, as well as |
repeat=“?” | providing a looping structure so that vXML |
cleardigits=“TRUE|FALSE” > | elements can be repeated a specific number |
Events | of times (e.g., a menu that plays an audio |
Elements | prompt four times before timing out.) |
</block> | |
<goto | This element will leap to another bank of |
value=“http://w.v.n/next.voxeo#block” | vXML code, whether it be in the same file |
submit=“all|*|x,y,z” | or another file. |
method=“put|post” /> | One example of Return is to implement |
<return/> | <goto> calls as a call stack. <Return> |
would return from a <goto> call. | |
<run | This runs/launches a vXML script at a URL |
value=“http://w.v.n/next.voxeo|#block” | or URI in a new session, then continues to |
submit=“all|*|x,y,z” | process this session |
method=“put|post” | |
newSessionID=“newID”/> | |
<sendevent value=“msg_call_answered” | This tag allows for one session to send a |
sessionID=“sss”/> | message to another session. |
TABLE 6B |
vXML Call Control Tags |
vXML CALL CONTROL TAG | |
Examples | DESCRIPTION |
<answer/> | This answers the call. |
<hangup/> | This informs the server to hangup |
<call value=“pstn:14079757500” | the call. This element allows for |
maxtime=“15s”/> | outbound calls to be created from |
<conference sessions=”sessionID1, | within a vXML script. This element |
sessionID2, sessionID3”/> | allows for multiple lines in |
separate sessions to be conferenced | |
together. | |
TABLE 6C |
vXML Media Tags |
VXML MEDIA TAG | |
Examples | DESCRIPTION |
<play... /> | <Playaudio> can be used to play an |
<playnumber | format=“say|read” | audio file and wait for a terminating |
value=“12345” | digit to be pressed. The element | |
termdigits=“*#” | must be located within a larger | |
cleardigits=“TRUE|FALSE”/> | <block> structure, which is used to | |
<playmoney | format=“???” | control the number of repetitions the |
value=“1.25” | audio is played before “timing out.” | |
termdigits=“*#” | Like the earlier example of | |
cleardigits=“TRUE|FALSE”/> | <getDigits>, <playaudio> can be | |
<playdate | format=“ddmmyyhhss” | implemented with event handlers to |
value=“1012990732” | properly recognize that the | |
termdigits=“*#” | <playaudio> command was halted | |
cleardigits=“TRUE|FALSE”/> | because a terminating digit was | |
<playchars | format=“?” | pressed by the user. |
value=“abcdefgh” | ||
termdigits=“*#” | ||
cleardigits=“TRUE|FALSE”/> | ||
<playtone | format=“?” | |
value=“2000hz+1000hz”\” | ||
termdigits=“*#” | ||
cleardigits=“TRUE|FALSE”/> | ||
<playaudio | format=“audio/msgsm” | |
value=“http://www.blahblah.com/ | ||
sample.vox” | ||
termdigits=“*#” | ||
cleardigits=“TRUE|FALSE”/> |
<recordaudio | format=“audio/msgsm” | Like its counterpart <playaudio>, |
value=“ftp://www.v.n/msg.wav” | this element must be contained | |
termdigits=“*#” | within a viable <block> structure | |
cleardigits=“TRUE|FALSE” | and utilize an event handler such | |
maxtime=“30s” | that one generated by a user action | |
ptime=“5s”/> | to control it. Its intended use is to |
leave voicemail messages, record | |
greetings, etc. In the example | |
above, the user would be allowed to | |
speak into the phone to record audio | |
for up to 30 seconds (with no more | |
than a 5 second pause anywhere | |
within), and could end the recording | |
segment by pressing either the * or | |
# key. The new audio file would | |
then be saved at www.v.n/msg.wav | |
in the audio/msgsm format. The | |
clearDigits attribute, again, is used | |
to ensure a “fresh” start during the | |
audio recording, in case one of the | |
terminating digits was pressed prior | |
to initiating the recording. |
<receivefax | format=“audio/tiff-f” | Receives a fax. |
value=“ftp://w.v.n/msg.tif” |
maxtime=“5m” | |
ptime=“30s” | |
maxpages=“30”/> |
<sendfax | format= “audio/tiff-f” | Sends a fax. |
value=“http://w.v.n/msg.tif” | ||
maxtime=“5m” | ||
ptime=“30s” | ||
maxpages=“30”/> |
<text | format=“?” | This is used to tell the application |
termdigits=“#” | gateway server to use a text-to- | |
cleardigits=“TRUE|FALSE”> | speech engine for reading the |
Text to read | enclosed text to the caller. |
</text> | |
<vcommand name=“id” value=“url|vocab- | This is used to tell the application |
grammar-string”> | gateway server to use a speech-to- |
text engine for voice recognition. | |
TABLE 6D |
vXML High Level Tags |
VXML HIGH LEVEL TAG | |
Examples | DESCRIPTION |
<menu label=“main_menu” | Menu is an element that is descended from |
repeat=“3” | a <block> element and a <playAudio> |
format=“audio/msgsm” | element.. In this embodiment, an |
value=“http://w.v.n/msg.wav” | <ontermdigit> event handler is used, to |
cleardigits=“TRUE|FALSE” | handle the event when a terminating digit |
termdigits=“567890*#” | has been pressed. It is designed to accept a |
maxtime=“15s” > | single digit input and then check for a |
Events | matching <onkey> event handler. This |
<onkey value=″1″> </onkey> | element is to allow easy-to-use menus, |
<onkey value=″2″> </onkey> | where one key press will move you through |
<onmaxtime value=″1|2|max”> | an application. In the example above (and |
</onmaxtime> | below), the audio file will be played 3 times |
<onhangup> | </onhangup> | before “timing out” and moving on in the |
</menu> | vXML code. |
<inputdigits label=“input_pin” | <InputDigits> is an element that is |
repeat=“3” | descended from a <block> element and a |
var=“pager_msg” | <getDigits> element. It combines the |
format=“audio/msgsm” | attributes of those two elements into a |
value=“http://w.v.n.msg.wav” | single element. Like the <menu> element |
termdigits=“#*” | above, <inputDigits> simplifies the process |
cleardigits=“TRUE|FALSE” | of making a function to gather digits from |
includetermdigit=“TRUE|FALSE” | the user. It will play a message (contained |
maxdigits=“4” | in the “value” attribute) and store the |
maxtime=“15s” | gathered information in the “var” attribute. |
ptime=“5s”> | In the example, the user has 15 seconds to |
Events | enter up to 4 digits (possibly for a PIN |
<oninputvalue value=“123”> | code), with a pause of no more than 5 |
</oninputvalue> | seconds between keystrokes. Either the # |
<oninputlength len=“3”> | or * key will end the input process, and the |
</oninputlength> | audio message/ |
<ontermdigit value=“#”> | before dropping out of the element (i.e., |
</ontermdigit> | timing out), and proceeding on to the rest of |
<onmaxdigits> | the vXML code. |
</onmaxdigits> | |
<onmaxtime value=“1|2|max”> | |
</onmaxtime> |
<onptime> | </onptime> |
<onhangup> | </onhangup> |
</inputdigits> |
<menu label=“main_menu” | Menu is an element that is descended from |
repeat=“3” | a <block> element and a <playAudio> |
format=“audio/msgsm” | element.. In this embodiment, an |
value=“http://w.v.n/msg.wav” | <ontermdigit> event handler is used, to |
cleardigits=“TRUE|FALSE” | handle the event when a terminating digit |
termdigits=“567890*#” | has been pressed. It is designed to accept a |
maxtime=“15s”> | single digit input and then check for a |
Events | matching <onkey> event handler. This |
<onkey value=″1″> | </onkey> | element is to allow easy-to-use menus, |
<onkey value=″2″> | </onkey> | where one key press will move you through |
<onmaxtime value=″1|2|max”> | an application. In the example above (and |
</onmaxtime> | below), the audio file will be played 3 times |
<onhangup> | </onhangup> | before “timing out” and moving on in the |
</menu> | vXML code. |
<inputdigits label=“input_pin” | <InputDigits> is an element that is |
repeat=“3” | descended from a <block> element and a |
var=“pager_msg” | <getDigits> element. It combines the |
format=“audio/msgsm” | attributes of those two elements into a |
value=“http://w.v.n.msg.wav” | single element. Like the <menu> element |
termdigits=“#*” | above, <inputDigits> simplifies the process |
cleardigits=“TRUE|FALSE” | of making a function to gather digits from |
includetermdigit=“TRUE|FALSE” | the user. It will play a message (contained |
maxdigits=“4” | in the “value” attribute) and store the |
maxtime=“15s” | gathered information in the “var” attribute. |
ptime=“5s”> | In the example, the user has 15 seconds to |
Events | enter up to 4 digits (possibly for a PIN |
<oninputvalue value=“123”> | code), with a pause of no more than 5 |
</oninputvalue> | seconds between keystrokes. Either the # |
<oninputlength len=“3”> | or * key will end the input process, and the |
</oninputlength> | audio message/ |
<ontermdigit value=“#”> | before dropping out of the element (i.e., |
</ontermdigit> | timing out), and proceeding on to the rest of |
<onmaxdigits> | the vXML code. |
</onmaxdigits> | |
<onmaxtime value=“1|2|max”> | |
</onmaxtime> |
<onptime> | </onptime> |
<onhangup> | </onhangup> |
</inputdigits> | |
MicroXML sent from the session manager to the vXML parser |
Request the parsing of a new file |
<ACT> |
<SID>24601</SID> |
<BUF> |
<?xml version=“1.0” encoding=“UTF-8”?> |
<voxeoxml> |
<assign var=“rootdir” value=“http://www.voxeo.com/”/> |
<block label=“1” repeat=“3”> |
<playaudio format=“audio/default” value=“$rootdir;greeting.vox”/> |
</block> |
</voxeoxml> |
</BUF> |
</ACT> |
Request the continued parsing of the same file |
<ACT> |
<SID>24601</SID> |
</ACT> |
Report a basic user event to the vXML parser |
<EVT> |
<SID>24601</SID> |
<ETP>termdigit</ETP> |
<EVL>#</EVL> |
</EVT> |
Report a user event with parameter(s) to the vXML parser |
<EVT> |
<SID>24601</SID> |
<ETP>termdigit</ETP> |
<EVL>#</EVL> |
<P00>assign=varname=value</P00> |
</EVT> |
MicroXML sent from the vXML parser to the session manager |
Action from parser |
<ACT> |
<TYP>playaudio</TYP> |
<PR1>format=audio/default,value=http://www.voxeo.com/greeting.vox |
</PR1> |
</ACT> |
Example vXML file |
<?xml version=“1.0” encoding=“UTF-8”?> | ||
<voxeoxml> | ||
<!-- This is a test file to exercise the voxeoXML parser --> | ||
<assign var=“audiodir” value=“http://www.voxeo.com/audio”/> | ||
<block label=“testlooping” repeat =“3”> | ||
<assign var=“foo” value=“$foo;bar”/> | ||
<playaudio | ||
format=“audio/msgsm” | ||
value=“$audiodir;/$foo;.vox” | ||
termdigits=“*” | ||
cleardigits=“true” | ||
/> | ||
</block> | ||
<hangup/> | ||
</voxeoxml> | ||
The resulting microXML |
Separate calls to parse are delimited by ‘----------------------------‘ |
---------------------------- |
<ACT> |
<SID>24601</SID> |
<TYP>cleardigits</TYP> |
</ACT> |
<ACT> |
<SID>24601</SID> |
<TYP>playaudio</TYP> |
<PR1>format=audio/msgsm,value=http://www.voxeo.com/audio/bar.vox, |
termdigits=*</PR1> |
</ACT> |
---------------------------- |
<ACT> |
<SID>24601</SID> |
<TYP>cleardigits</TYP> |
</ACT> |
<ACT> |
<SID>24601</SID> |
<TYP>playaudio</TYP> |
<PR1>format=audio/msgsm,value=http://www.voxeo.com/audio/barbar.- |
vox,termdigits=*</PR1> |
</ACT> |
---------------------------- |
<ACT> |
<SID>24601</SID> |
<TYP>cleardigits</TYP> |
</ACT> |
<ACT> |
<SID>24601</SID> |
<TYP>playaudio</TYP> |
<PR1>format=audio/msgsm,value=http://www.voxeo.com/audio/ |
barbarbar.vox,termdigits=*</PR1> |
</ACT> |
---------------------------- |
<ACT> |
<SID>24601</SID> |
<TYP>hangup</TYP> |
<PR1></PR1> |
</ACT> |
---------------------------- |
<ACT> |
<SID>24601</SID> |
<TYP>EOF</TYP> |
<PR1></PR1> |
</ACT> |
-
- STEP 710: In a networked computer telephony system operating under the Internet Protocol, providing a first set of application gateway centers in a hosted facility and a predetermined second set of application gateway centers at a subscriber's premise for receiving and processing telephone calls.
- STEP 720: Providing a directory for routing a telephone call identified by a telephone number to an application gateway center from either the first set or the second set, the directory designating an application gateway center as a function of the telephone number, predefined routing rules and operating capacities of individual application gateway centers.
- STEP 730: Updating the directory at predefined time intervals on the operating capacities of the individual application gateway centers.
- STEP 740: Providing a user interface for a user to update user-specified routing rules in the directory.
- STEP 742-759, any one of them is optional (see
FIGS. 18A-18J .) - STEP 760: Routing the telephone call according to the directory.
Claims (27)
Priority Applications (6)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
US12/195,298 US8355394B2 (en) | 2007-08-21 | 2008-08-20 | System and method for dynamic telephony resource allocation between premise and hosted facilities |
PCT/US2008/073892 WO2009026452A1 (en) | 2007-08-21 | 2008-08-21 | Dynamic routing of telephone calls to one out of a plurality of computer telephony servers comprising premises- and network-based servers |
EP08798394A EP2183907A1 (en) | 2007-08-21 | 2008-08-21 | Dynamic routing of telephone calls to one out of a plurality of computer telephony servers comprising premises- and network-based servers |
AU2008288853A AU2008288853B2 (en) | 2007-08-21 | 2008-08-21 | Dynamic routing of telephone calls to one out of a plurality of computer telephony servers comprising premises- and network-based servers |
CA2697058A CA2697058C (en) | 2007-08-21 | 2008-08-21 | Dynamic routing of telephone calls to one out of a plurality of computer telephony servers comprising premises- and network-based servers |
US13/739,983 US8891511B2 (en) | 2007-08-21 | 2013-01-11 | System and method for dynamic telephony resource allocation between premise and hosted facilities |
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US95715107P | 2007-08-21 | 2007-08-21 | |
US12/195,298 US8355394B2 (en) | 2007-08-21 | 2008-08-20 | System and method for dynamic telephony resource allocation between premise and hosted facilities |
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US13/739,983 Continuation US8891511B2 (en) | 2007-08-21 | 2013-01-11 | System and method for dynamic telephony resource allocation between premise and hosted facilities |
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US20090052437A1 US20090052437A1 (en) | 2009-02-26 |
US8355394B2 true US8355394B2 (en) | 2013-01-15 |
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US12/195,298 Active 2031-08-19 US8355394B2 (en) | 2007-08-21 | 2008-08-20 | System and method for dynamic telephony resource allocation between premise and hosted facilities |
US13/739,983 Expired - Fee Related US8891511B2 (en) | 2007-08-21 | 2013-01-11 | System and method for dynamic telephony resource allocation between premise and hosted facilities |
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US13/739,983 Expired - Fee Related US8891511B2 (en) | 2007-08-21 | 2013-01-11 | System and method for dynamic telephony resource allocation between premise and hosted facilities |
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---|---|
US (2) | US8355394B2 (en) |
EP (1) | EP2183907A1 (en) |
AU (1) | AU2008288853B2 (en) |
CA (1) | CA2697058C (en) |
WO (1) | WO2009026452A1 (en) |
Cited By (51)
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US20100150139A1 (en) * | 2008-10-01 | 2010-06-17 | Jeffrey Lawson | Telephony Web Event System and Method |
US20130128880A1 (en) * | 2007-08-21 | 2013-05-23 | Voxeo Corporation | System and Method for Dynamic Telephony Resource Allocation Between Premise and Hosted Facilities |
US8995641B2 (en) | 2009-03-02 | 2015-03-31 | Twilio, Inc. | Method and system for a multitenancy telephone network |
US9001666B2 (en) | 2013-03-15 | 2015-04-07 | Twilio, Inc. | System and method for improving routing in a distributed communication platform |
US9137127B2 (en) | 2013-09-17 | 2015-09-15 | Twilio, Inc. | System and method for providing communication platform metadata |
US9160696B2 (en) | 2013-06-19 | 2015-10-13 | Twilio, Inc. | System for transforming media resource into destination device compatible messaging format |
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CA2697058C (en) | 2016-08-16 |
AU2008288853B2 (en) | 2014-03-20 |
AU2008288853A1 (en) | 2009-02-26 |
WO2009026452A1 (en) | 2009-02-26 |
US20090052437A1 (en) | 2009-02-26 |
EP2183907A1 (en) | 2010-05-12 |
CA2697058A1 (en) | 2009-02-26 |
US20130128880A1 (en) | 2013-05-23 |
US8891511B2 (en) | 2014-11-18 |
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